TWI734176B - Acoustic processor having low latency - Google Patents

Abstract

An audio system having low latency includes a digital audio processor as well as sensor inputs coupled to the processor. The sensor inputs may be microphone inputs. The audio processor operates at the same frequency as the sensor inputs, which is typically much higher than an audio signal provided to the audio processor. In some aspects the audio processor operates as a noise cancellation processor and does not include an audio input.

Description

Acoustic wave processor with low latent time

This case is aimed at sonic processing, and more specifically, it is aimed at a reconfigurable sonic processor, which can run in real time or near real time.

Generally speaking, the presence of noise in the listening environment almost always endangers the experience of listening to audio through headphones. For example, in the cabin of an aircraft, in addition to audio programs, noise from the aircraft generates unwanted sound waves (that is, noise) and is transmitted to the listener's ears. Other examples include computer and air-conditioning noise in offices or houses, vehicle and passenger noise in public or private transportation, or other noisy environments.

In trying to reduce the amount of noise received by the listener, two main methods of noise reduction have been developed: passive noise reduction and active noise cancellation. Passive noise reduction refers to reducing the noise caused by placing a physical barrier (which is usually a headset or earplugs) between the ear cavity and the noisy external environment. The amount of noise reduced depends on the quality of the barrier. Generally speaking, headphones with higher quality noise reduction provide higher passive noise reduction. However, large and heavy headphones can be uncomfortable to wear for long periods of time. For a given headset, passive noise reduction is better to reduce higher frequency noise, while low frequencies can still pass through the passive noise reduction system.

Active noise reduction system is also called Active Noise Cancellation (ANC), which refers to the reduction of noise by playing anti-noise signals through the speakers of the headset. The anti-noise signal is generated as a negative approximation of the noise signal in the ear cavity without ANC. When the anti-noise signal is combined, the noise signal is neutralized.

In the normal noise cancellation process, one or more sensors (such as microphones) monitor the surrounding noise or the noise in the earphone barrel of the headset in real time, and then the system generates an anti-noise signal from the surrounding or residual noise. The anti-noise signal can be generated differently, depending on many factors, such as the physical shape and size of the ANC system (such as headphones...), the frequency response of sensors and transducers (such as speakers) , The latent time of the transducer at various frequencies, the sensitivity of the sensor, the placement of the transducer and the sensor. The above factors change between different sensors and transducers (such as headphones) and even between two earphone tubes of the same headphone system, which means that optimal filtering of noise is generated. The design of the device has also changed.

Dealing with the latent time in the anti-noise signal prevents the active noise cancellation system from operating efficiently. For example, digitizing and processing a sensor signal at a rate common to audio processing (such as 44.1 kHz or 48 kHz) introduces a large latency. Because the performance of a sound wave processor (such as ANC) depends on the ability to detect noise and generate an anti-noise signal to eliminate noise quickly enough in time, a large latent time is harmful to the sound wave noise cancellation process.

The embodiments of the present invention solve these limitations of the above-mentioned prior art.

In view of the above-mentioned shortcomings of the conventional technology, the purpose of the present invention is to provide an audio system to solve the problem that the audio processing of the conventional technology introduces a large latent time so as to suppress the sound wave noise cancellation processing.

To achieve the above objective, an audio system provided by the present invention mainly includes a sensor and a digital audio processor. Wherein, the sensor generates a digital sensor signal at a first rate higher than 50 kHz. The digital audio processor operates at the first rate, and has a first input to receive input sound signals at the first rate, a second input to receive the sensor signals at the first rate, and Has output.

In order to solve the above technical problems, a reconfigurable noise cancellation system is further proposed, which includes: an input, an interposer, at least one sensor; and a digital audio processor. Among them, the input receives a digital audio signal at the first rate. The interpolator changes the sampling rate of the digital audio signal from the first rate to a second rate higher than the first rate. The at least one sensor generates a sensor signal at the second rate. The digital audio processor is coupled to the interposer and the sensor, and the reconfigurable audio processor operates at the second rate. The digital audio processor further includes: multiple programmable filters, multiple controllable gain stages, adders and audio output. Wherein, at least some of the plurality of controllable gain stages are respectively coupled to at least some of the plurality of programmable filters. The adder is structured to combine the output of the one or more gain stages. The audio output is coupled to at least one of the adders to transmit an output audio signal modified from the input audio signal.

According to a preferred embodiment of the present invention, a method of operating an audio system includes: operating a digital audio processor at a first rate of 50 kHz or higher, and receiving the digital audio processor at the first rate The digital input audio signal of the digital audio processor receives the digital sensor signal with the first rate, and the digital audio processor running at the first rate combines the digital input audio signal And the signal derived from the digital sensor signal, and process the digital input audio signal, and output the processed digital input audio signal at the output.

According to another preferred embodiment of the present invention, a method of operating a reconfigurable noise cancellation processor includes: receiving a sound signal through an audio input at a first frequency, and transmitting one or more signals at the first frequency. A sensor input receives one or more sensor signals for monitoring the environment, and the filter parameter section of a plurality of programmable filters is constructed in the reconstructable noise canceling processor, and the filter parameter section of the programmable filter is constructed in the reconstructable The constructed noise cancellation processor constructs multiple controllable gain stages, and mixes the selected outputs of the multiple controllable gain stages with the sound signal at the first frequency to produce a modified sound Signal output. Wherein, at least some of the plurality of controllable gain stages are respectively coupled to at least some of the plurality of programmable filters.

Regarding the detailed structure and operation method of the audio system, the reconfigurable noise cancellation system, the method of operating the audio system, and the method of operating the reconfigurable noise cancellation processor proposed by the present invention, the following examples will be listed in conjunction with the drawings , So that those with ordinary knowledge in the field of the present invention can realize the specific embodiments of the present invention.

The embodiment of the present invention is directed to a digital acoustic wave processor, such as a reconfigurable acoustic wave processor (RAP), which is used in an audio system using digital sensor input.

There are three main types of active noise cancellation (ANC), which are distinguished based on the placement of the sensors or microphones in the system. In the feed-forward ANC, the sensor senses the surrounding noise, but imperceptibly senses the signal generated by the transducer (such as a speaker). Such a system is demonstrated in Figure 1. Please refer to FIG. 1, the feedforward ANC system 10 includes a sensor 12, which senses the surrounding noise but does not monitor the signal directly from the transducer 14. The output from the sensor 12 is filtered in the feedforward filter 16, and the filter output is coupled to the feedforward mixer 18, where the filtered signal is mixed with the input audio signal. The filtered signal from the filter 16 is an anti-noise signal generated from the output of the sensor 12. When the anti-noise signal is mixed with the desired signal in the mixer 18, the output of the transducer 14 is a combination of the input signal that filters the anti-noise signal, and the noise is less than if the anti-noise signal is not produced The noise.

In the feedback ANC, the position of the sensor is to sense the overall sound signal present in the ear cavity. In other words, the sensor senses the sum of the surrounding noise and the audio played by the transducer. An example of such a system is shown in Figure 2. Please refer to FIG. 2, in the feedback ANC system 20, the sensor 32 directly monitors the output from the transducer 24. The output from the sensor 32 is mixed with the audio input signal in the feedback mixer 30, and then the combined signal is sent to the feedback filter 34, where the combined signal is filtered to generate an anti-noise signal. The anti-noise signal from the filter 34 is mixed with the original sound signal in the mixer 28, and the combined output is then fed to the transducer 24. The feedback ANC system 20 also reduces the noise heard by the listener of the speaker 24.

The combined feedforward and feedback ANC system uses two or more sensors. The first position of the sensor is in the feedforward path as shown in Figure 1, and the second position of the sensor is in Figure 2. The feedback path shown. The combined feedforward and feedback ANC system 40 is illustrated in FIG. 3 and includes sensors at positions 42 and 52 and one or more transducers at position 44, as shown in FIG. 3. The signals sensed from the feedback sensor(s) at position 52 are mixed in the feedback mixer 50, and the combined signal is filtered by the feedback filter 54. Likewise, the signal sensed from the feedforward sensor(s) at position 42 is filtered in the feedforward filter 46, and the filtered signal is combined with the incoming audio signal in the feedforward mixer 48. The output of the transducer(s) at position 44 has reduced noise through filtering and mixing operations.

Although existing systems use fixed topologies and filters, embodiments of the present invention use alternative systems to cover many different applications, as detailed below.

The typical audio processing rate is 44.1 kHz or 48 kHz, which is based on the typical frequency range of human hearing. At these sampling rates, the sampling time is around 20 microseconds. Digitization and filtering in the ANC system take multiple samples unchanged. At these rates, the delay caused is on the order of hundreds of microseconds. Because any delay in processing will degrade the generation of anti-noise signals, this significantly reduces ANC performance. This often manifests itself as limiting the maximum noise frequency that can be eliminated.

FIG. 4 is a block diagram of the audio system 100, which includes a low-latency or ultra-low-latency acoustic wave processor. In some embodiments, the acoustic wave processor may be reconfigurable, and is referred to as a reconfigurable audio processor (RAP) 150. The audio system of Figure 4 is roughly divided into three parts: the analog part 102, the digital part 104 that runs at the rate of the analog-to-digital converter (ADC), and the digital part that runs at the standard audio sampling rate (for example, 44.1 or 48 kHz). Part 106. These parts can also be called domains.

The analog part 102 does not require a clock, and typically, the signal in this part is generally a continuous analog signal. For example, the transducer or speaker 110 can generate analog sound signals from a headset or other speakers, for example. For example, a sensor such as the digital microphone 112 automatically generates a digital output from an analog input signal; meanwhile, a standard analog sensor such as the microphone 114 can be combined with the ADC 124 to generate a digital signal from the analog sensor 114. The sensor 116 such as a microphone can be placed in the feedback position and coupled to the ADC 126. For example, ADCs 124, 126 can use sigma-delta processing. In other embodiments, the ADCs 124 and 126 may be pulse code modulation (PCM) or continuous approach register (SAR) types. A single sensor 112, 114, 116 can be used for multiple purposes, for example, for example, sampling the surrounding noise while also serving as an input microphone for the phone. There may be one or more filters 128 to filter the output from the ADC 124, 126, but not all embodiments are required.

A digital signal processor (DSP) 130 or other audio source operates in the digital part 106, and the operating frequency is at a standard audio sampling rate. Generally speaking, the operating frequency of the digital part 106 of the audio system 100 can be 44.1 or 48 kHz.

Relatively speaking, the operating frequency of the digital part 104 can be operated from a rate of 50 kilohertz to a rate of 100 megahertz, and is preferably in the range of, for example, 2-100 megahertz. In some embodiments, the digital part 104 can operate at 50 kHz, 96 kHz, in the range of hundreds of thousands of hertz, at frequencies in the low-megahertz range (for example, 1 to 6), and in the range of tens of megahertz. The range (for example, 10-20 MHz), up to 100 MHz. In the embodiment of the present invention, each component in a particular field operates at the frequency of the field. For example, referring to FIG. 4, the ADCs 124 and 126 operate at the same frequency as the audio processor or RAP 150. This is very different from previous systems, which typically use a destruction filter to downsample the sensor signal before processing it in the audio processor.

The inserter 140 converts the audio signal from the DSP 130 (for example, operating at 48 kHz) into an audio signal operating at 3 MHz or 6 MHz as an input signal to the RAP 150. In contrast, the destroyer 144 (which does not need to be present in all audio systems 100) converts the signal from the RAP 150 (for example, at 3 or 6 MHz) into the operating frequency of the digital part 106. The resulting latency of RAP 150 is extremely low, for example, less than 2.5 microseconds, and preferably less than 0.5 microseconds, because the rate at which RAP 150 processes signals is the same as the rate at which sensors or microphones 112, 114, 116 generate them , Regardless of whether the sensor is a digital microphone or whether the sensor signal is converted to a digital signal by the ADC 124, 126.

As described in more detail below, the RAP 150 controls, for example, the acoustic signal emitted from the transducer 110 in real time. As mentioned above, RAP 150 is structured to operate on raw sensor samples from microphones 112, 114 and/or 116 without any intermediate processing, like destroying filters or other sample rate converters. This allows the RAP 150 to respond to the microphone signal with zero or nearly zero arithmetic delay, which can implement real-time audio processing algorithms. The effect of using real-time sensor sampling is to eliminate the delay from the destruction filter of the previous system, which in turn greatly increases the responsiveness of the control loop.

The sampling rate of the digital part 104 can be changed according to the sampling rate of the digital sensor 112 or the ADC 124 coupled to the analog sensor 114. There is a linear trade-off between the sampling rate and the throughput of each sample.

FIG. 5 is a functional block diagram of an example of a reconfigurable acoustic wave processor (RAP) 250, which may be an embodiment of the RAP 150 in FIG. 4. The RAP 250 in FIG. 5 includes six chains of double quad filters BQ0 to BQ6, and their functions are described as follows. The double quad filter is well known for electrical processing (especially audio processing). The double quad filter typically includes 2 zeros and 2 poles. Each of the double quad chains BQ0~BQ6 includes a series of double quad filters. In some embodiments, the chains BQ0~BQ6 may include 4, 6, 8, 12 or 16 double quad filters in series, and 8 are preferred. The double-quad filter chain BQ0~BQ6 is programmable, so the filter value can be changed according to the desired implementation. They can also be set to pass or unitary settings, which means that they do not significantly affect the signals passing through them.

Connected to each double quad filter chain BQ0~BQ6 are gain units M0~M6, and there are additional gain units M7, the purpose of which is described as follows. The programmable gain unit M0-M7 is that the amount of gain generated between its input and output is controllable. The output of the special double-four filter chain BQ0~BQ6 can be controlled by its coupled gain unit M0~M6. Setting the gain of any gain unit M0~M6 to zero effectively closes the special circuit branch. Although there is no need to strictly maintain a one-to-one relationship between the double-quad filter chain and the gain unit, maintaining this relationship provides flexibility in setting RAP. The RAP 250 of Figure 5 shows a single channel. For two or more channels, for example for stereo processing, additional hardware is used.

By programming the special filter coefficients in the double quad filter chain BQ0~BQ6 and the special gain values in the gain units M0~M6, different audio applications can be performed in the RAP 250, such as audio noise cancellation, as described below .

Also coupled to the RAP 250 may include inputs from digital sensors 212, 214 (which may be microphones), a destroyer 218, and an inserter 220. Either or both of the sensor inputs 212, 214 can be generated by coupling an analog microphone to the ADC. The operations of the destroyer 218 and the inserter 220 are as described with reference to FIG. 4.

Operationally, the RAP 250 accepts inputs from the sensors 212 in the dual quad filter chains BQ0 and BQ3, and accepts inputs from the sensors 214 in the dual quad filter chains BQ1 and BQ5. Receive audio signals in the double quad filter chain BQ2 and BQ6. In some embodiments, the audio signal is not strictly necessary. For example, in noise cancelling headsets used in hunters or industries, there may be no sound signal.

Before the processed audio signal is finally combined with the unprocessed audio signal from the inserter 220 in the combiner A2, the gain unit M7 can be used as a controllable gain for the processed audio signal. The gain unit M7 can be controlled to gradually increase its gain, so that noise cancellation or other processing can be gradually added to the unprocessed sound signal to eliminate pops or other rapid changes in the output sound signal, which can be undesirable to the listener. comfortable.

The adder or combiner A0, A1, and A2 combine the intermediate signal output from the double quad filter chain, as shown in Figure 5.

In one embodiment, the RAP 250 operates at 49.152 MHz, which is the standard rate for audio processing. The input sampling rate is generally 3.072 million times per second, and the filtering part can also operate at the same rate.

An intuitive example of the operation of the RAP 250 is a pure audio processor without using input from either of the sensors 212, 214. In this example, the gain unit M7 is set to 0 (that is, off), and the audio signal from the inserter is filtered by the double quad filter chain BQ6. The gain control unit M6 controls the output signal level of the filtered sound signal, and this signal is sent to the transducer 210, which can be a speaker or other transducer output.

In a more complicated example, the RAP 250 can be constructed as a feedforward/feedback ANC, which has the same functionality as the feedforward and feedback ANC circuits demonstrated in FIG. 3. Figure 6 demonstrates how the RAP 250 is set to this configuration. In this configuration, the gain units M0 and M5 are set to 0, so they are not shown in FIG. 6. The gain units M2, M4, M6, and M7 are set to 1. The gain units M1 and M3 are set to -1, which means that their output is subtracted. The double-quad filter chains BQ1, BQ2, and BQ6 are set as pass settings. Please refer to FIGS. 3 and 6, the double quad filter chain BQ3 has the role of the feedforward filter 46, and the double quad filter chain BQ4 has the role of the feedback filter 54.

By constructing the RAP 250, especially the gain units M0~M7 and the double quad filter chains BQ1~BQ6, the RAP can be constructed to perform most any type of audio processing. For example, the RAP 250 can be constructed as an ANC processor for active noise cancelling headsets to form feedback, feedforward, or combined feedforward feedback configurations. The RAP 250 can be used for active noise cancellation of the phone handset by using the input from the handset microphone and generating audio output for one or more speakers of the handset. RAP 250 can further enhance the input audio signal while eliminating noise. RAP 250 can also be used to enhance the surrounding sound by accepting the surrounding sound input by a certain microphone, modifying the sound through one or more double quad filter chains, setting the appropriate gain level, and then outputting the modified surrounding signal. .

In practice, the RAP 250 in FIG. 6 or the RAP 250 in FIG. 5 includes a function, process, or operation for modifying the input of a sound signal. In practice, these functions can be implemented by specially formed hardware circuits into programming functions that operate on general-purpose or special-purpose processors (such as digital signal processors (DSP)); or can be implemented in field programmable gate arrays (FPGA) or programmable logic device (PLD). There may also be other variations.

FIG. 7 is a functional block diagram of components of the exemplary re-constructable acoustic wave processor of FIG. 4 according to an embodiment of the present invention. As shown in FIG. 7, the RAP 350 includes a dual-four engine 310 and a multiplication accumulator 320. The multiplication accumulator 320 implements all the multipliers and adders in the functional block diagrams of FIGS. 5 and 6. In one embodiment, there are seven multiply-add operations for each sample. The dual four engine 310 includes inputs from one or more sensors (such as microphones) and input of audio signals to be processed. The dual four engines can also accept input from the output of the multiplying accumulator. The input from the sensor is timed at the same rate as the double quad engine. In other words, the sensor input can be processed without any destruction or rate reduction. The size of the dual quad engine 310 can be made to operate on 16 dual quad filters. The double quad descriptor section 330 includes filtering values to implement the double quad filter chain, and the double quad state memory 332 is a memory that stores intermediate values during the double quad processing. The gain table 322 stores the values used for the gain unit, and the feathering control (for example, provided by the gain unit M7 in FIG. 5) is separately provided by the feathering control 334. The RAP 350 is programmed and constructed by writing special values into the double four descriptor 330 and the gain table 322, as shown in FIG. 7.

By using this programmable technique, filters can be selected to enhance rather than reduce specific sounds or noise. For example, the parameter of the double quad-chain filter is not selected for its ability to reduce the sound sensed by the special microphone. As mentioned above, the parameter can instead be selected to enhance the special sound. For example, someone may be using a noise-cancelling headset in a noisy work environment with all kinds of rumbling machinery, but still wants to be able to speak to a colleague without removing the noise-cancelling headset. Using adaptive filter coefficients, when the microphone detects noise in the voice band, different parameters can be automatically loaded into the RAP system that enhances the voice of colleagues. Therefore, the listener will have a noise-canceling headset that can adapt to enhance special sounds. For example, you can enhance sounds such as voice, TV audio, and traffic. When such a voice disappears, for example, when a colleague stops talking, the standard filter coefficients may be dynamically loaded into the filter of the RAP system again.

The embodiments of the present invention can be incorporated into integrated circuits, such as sound processing circuits or other audio circuits. Furthermore, integrated circuits can be used in audio devices, such as headsets, mobile phones, portable computing devices, sound sticks, audio platforms, amplifiers, speakers...

After describing and demonstrating the principles of the present invention through the exemplary embodiments, it will be recognized that the exemplary embodiments can be modified in configuration and details without departing from these principles, and can be combined in any desired manner. At the same time, although the previous discussion has focused on specific embodiments, other configurations are also considered.

In particular, even if expressions such as “embodiment according to the present invention” or similar are used herein, these terms mean the possibility of roughly referring to the embodiment, and are not intended to limit the present invention to a particular implementation configuration. As used herein, these terms can refer to the same or different embodiments, which can be combined into other embodiments.

Therefore, in view of the various modifications of the embodiments described herein, the [Embodiment Mode] and its accompanying materials are only intended for demonstration, and should not be considered as limiting the scope of the present invention.

10‧‧‧Feed-forward active noise cancellation (ANC) system 12‧‧‧Sensor 14‧‧‧Transducer 16‧‧‧Feedforward filter 18‧‧‧Feedforward mixer 20‧‧‧Feedback ANC System 24‧‧‧Transducer 28‧‧‧Mixer 30‧‧‧Feedback Mixer 32‧‧‧Sensor 34‧‧‧Feedback filter 40‧‧‧Combined feedforward and feedback ANC system 42‧‧‧Sensor 44‧‧‧Transducer 46‧‧‧Feedforward filter 48‧‧‧Feedforward mixer 50‧‧‧Feedback Mixer 52‧‧‧Sensor 54‧‧‧Feedback filter 100‧‧‧Audio System 102‧‧‧Analogous part 104‧‧‧Digital part running at the rate of analog-to-digital converter (ADC) 106‧‧‧Digital part running at standard audio sampling rate 110‧‧‧Transducer or speaker 112‧‧‧Digital Sensor 114‧‧‧Analog Sensor 116‧‧‧Sensor 124, 126‧‧‧ADC 128‧‧‧Filter 130‧‧‧Digital Signal Processor (DSP) 140‧‧‧Insertor 144‧‧‧Destroyer 150‧‧‧Reconfigurable Audio Processor (RAP) 210‧‧‧Transducer 212, 214‧‧‧Digital Sensor 218‧‧‧Destroyer 220‧‧‧Insertor 250‧‧‧RAP 310‧‧‧Double Four Engine 320‧‧‧Multiplication Accumulator 322‧‧‧Gain table 330‧‧‧Double Four Descriptor Section 332‧‧‧Dual four-state memory 334‧‧‧Feathering control 350‧‧‧RAP A0, A1, A2‧‧‧Adder or combiner BQ0~BQ6‧‧‧Double Quad Filter M0~M7‧‧‧Gain unit

Figure 1 is a circuit diagram demonstrating the conventional topology of feedforward active noise cancellation.

Figure 2 is a circuit diagram demonstrating the conventional topology of feedback active noise cancellation.

Figure 3 is a circuit diagram demonstrating the conventional topology that combines feedforward and feedback active noise cancellation.

FIG. 4 is a block diagram of an audio system, which includes a reconfigurable acoustic wave processor according to an embodiment of the present invention.

FIG. 5 is a functional block diagram of the exemplary reconfigurable acoustic wave processor of FIG. 4. FIG.

Fig. 6 is a block diagram of the reconfigurable acoustic wave processor of Fig. 4, which is constructed to implement combined feedforward and feedback active noise cancellation operations.

FIG. 7 is a functional block diagram of components of the exemplary re-constructable acoustic wave processor of FIG. 4 according to an embodiment of the present invention.

100‧‧‧Audio System

102‧‧‧Analogous part

104‧‧‧Digital part running at the rate of analog-to-digital converter (ADC)

106‧‧‧Digital part running at standard audio sampling rate

110‧‧‧Transducer or speaker

112‧‧‧Digital Sensor

114‧‧‧Analog Sensor

116‧‧‧Sensor

124, 126‧‧‧ADC

128‧‧‧Filter

130‧‧‧Digital Signal Processor (DSP)

140‧‧‧Insertor

144‧‧‧Destroyer

150‧‧‧Reconfigurable Audio Processor (RAP)

Claims (18)

An audio system, comprising: an analog part with a plurality of input sensors and at least one output device, the analog part does not require a clock; a first digital part with the plurality of input sensors and the At least one output device is electrically coupled to a first processor, the first digital portion runs at a first clock frequency, the first processor includes a plurality of programmable bi-quadratic filter chains, the plurality of A programmable double-quad filter chain is structured to receive signals from the plurality of input sensors and perform audio processing on the received signals; and a second digital part having the same A processor is electrically coupled to a second processor, and the second digital portion runs at a second clock frequency lower than the first clock frequency. For example, the audio system of claim 1, wherein the first processor further includes a plurality of controllable gain stages, at least some of the plurality of controllable gain stages are electrically coupled to the plurality of programmable double-quad filters At least some in the chain. For example, the audio system of claim 2 further includes an adder to combine the output of one or more gain stages. Such as the audio system of claim 1, wherein the first processor is a reconfigurable (reconfigurable) audio processor. For example, the audio system of claim 4, wherein the second processor is a digital signal processor. Such as the audio system of claim 1, wherein the plurality of input sensors include at least one microphone. Such as the audio system of claim 1, wherein the at least one output device includes a transducer. For example, the audio system of claim 1, further comprising an interpolator electrically coupled between the first processor and the second processor, wherein the interposer is configured to transfer a signal from the first processor The second clock frequency is converted to the first clock frequency. For example, the audio system of claim 8, which further includes a decimator electrically coupled between the first processor and the second processor, wherein the decimator is configured to send a signal from the first processor A clock frequency is converted to the second clock frequency. Such as the audio system of claim 1, wherein the first clock frequency is in the range of 4-100 megahertz (MHz). Such as the audio system of claim 1, wherein the second clock frequency is a frequency of approximately 48 kilohertz (KHz) or slower. A method of operating an audio system, comprising: a digital audio processor operating at a sampling frequency of an analog-to-digital converter; the digital audio processor receiving a digital input audio signal having the sampling frequency of the analog-to-digital converter The digital audio processor receives a digital sensor signal with a sampling frequency of the analog-to-digital converter via the analog-to-digital converter without the need for an intermediate sampling rate converter; the digital audio processor combines the digital input An audio signal and a signal derived from the digital sensor signal to perform noise cancellation on the digital input audio signal; and an output device that outputs the processed digital input audio signal. For example, the method of operating an audio system of claim 12, further comprising the digital audio processor controllably adjusting a gain of the signal derived from the digital sensor signal to controllably adjust the gain during the operation of the audio system A level of noise cancellation performed on the digital input audio signal. For example, the method of operating an audio system of claim 12, wherein receiving the digital input audio signal includes receiving a microphone signal at the sampling frequency of the analog-to-digital converter. For example, the method of operating an audio system of claim 12, wherein the output device is a transducer. For example, the method of operating an audio system of claim 12, wherein the sampling frequency of the analog-to-digital converter is 50 kilohertz (KHz) or higher. For example, the method of operating an audio system of claim 12, which includes: receiving the digital audio signal at an audio sampling frequency; converting the digital input audio signal from the audio sampling frequency to the analog-to-digital converter sampling frequency; and The analog-to-digital converter sampling frequency sends the digital audio signal to the digital audio processor. For example, the method of operating an audio system in claim 17, wherein the audio sampling frequency is 100 kilohertz (KHz) or lower.

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